• Journal of the Korea Society of Computer and Information
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• v.29 no.6
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• pp.89-100
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• 2024

The primary goal of software testing is to identify and correct errors within software. A key challenge in this process is error masking, where errors disappear internally before reaching the output. This paper investigates the causes and characteristics of error masking, which complicates software testing. The study involved injecting artificial errors into three software programs to examine the extent of error masking by various test cases and to explore the underlying reasons. The experiment yielded four major findings. First, about 50% of the error masking occurred because the errors were not executed. Second, among various operators, logical and arithmetic operators masked errors less frequently, while relational and temporal operators tended to mask errors more extensively. Third, certain test cases demonstrated exceptional effectiveness in propagating errors to the output. Fourth, the type of error injected influenced the masking effect.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2002.12a
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• pp.353-356
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• 2002

본 논문은 카오스 신호를 이용하여 안전한 음성신호의 전송을 위한 아날로그 비화통신 시스템의 성능분석에 관한 연구이다. 기존의 카오스 동기화 및 카오스 변조통신 알고리즘을 개선하여 실제 통신환경에서 발생하는 다양한 조건들을 적용하여 음성신호의 복원능력을 모의실험으로 분석하였다. 일반적인 PC 제어기법과 제안한 피드백 마스킹 기법을 사용하여 송신단에서 음성신호를 카오스 신호로 마스킹하여 변조하고, 통신채널에 잡음신호를 추가하여 전송하였다. 수신단에서는 카오스 응답시스템을 이용하여 음성신호를 복조하고, 복원성능을 계산하기 위하여 아날로그 복원 에러신호의 평균전력을 제안하여 계산하였다. 실험결과 마스킹 정도, 파라미터들의 민감성, 채널잡음 등에 대하여 PC 제어기법보다 피드백 제어기법의 복원성능이 우수함을 확인할 수 있었다. 또한 로렌쯔 카오스 시스템을 비화통신시스템에 사용할 경우 파라미터들의 조합으로 암호키를 구성해야 하므로 키값들의 선정에 기준이 되는 파라미터 변화율에 대응하는 복원에러율의 관계를 실험 값으로 구하였다.

• Journal of the Korean Institute of Intelligent Systems
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• v.13 no.6
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• pp.698-703
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• 2003

This paper presents analog secure communication system about safe speech transmission using chaotic signals. We applied various conditions that happen in actuality communication environment modifying chaotic synchronization and chaotic communication schemes and analyzed restoration performance of speech signal to computer simulation. In transmitter, we made the chaotic masking signal which is added voice signal to chaotic signal using PC(Pecora ＆ Carroll) and SFB(self-feedback) control techniques and transmitted encryption signal to noisy communication channel And in order to calculate the degree of restoration performance, we proposed the definition of analog average power of recovered error signals in receiver chaotic system. The simulation results show that feedback control techniques can certify that restoration performance is superior to quantitative data than PC method about masking degree, susceptibility of parameters and channel noise. We experimentally computed the table of relation of parameter fluxion to restoration error rate which is applied the encryption key values to the chaotic secure communication.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.68-77
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• 2000

Line special frequency (LSF) parameters have been widely used in low bit-rate speech coding due to their efficiency for representing the short-time speech spectrum. In this paper, a new distance measure based on the masking properties of human ear is proposed for quantizing LSF parameters whereas most conventional quantization methods are based on the weighted Euclidean distance measure. The proposed method derives the perceptual distance measure from the definition of noise-to-mask ratio (NMR) which has high correspondence with the actual distortion received in the human ear and uses it for quantizing LSF parameters. In addition, we propose an adaptive bit allocation scheme, which allocates minimal bits to LSF parameters maintaining the perceptual transparency of given speech frame for reducing the average bit-rates. For the performance evaluation, we has shown the ratio of perceptually transparent frames and the corresponding average bit-rates for the conventional and proposed methods. By jointly combining the proposed distance measure and adaptive bit allocation scheme, the proposed system requires only 770 bps for obtaining 95.5% perceptually transparent frames, while the conventional systems produce 89.9% at even 1800 bps.